ccm7 sip trunk

© 2010 Cisco Systems, Inc. All rights reserved. Important notices, privacy statements, and trademarks of Cisco Systems, Inc. can be found on cisco.com

Note: Results and technical information in this document were obtained through joint testing of Verizon business and Cisco engineers within Verizon laboratories. of 49

EDCS# 846444 Rev # 2

Application Note

Verizon IP Trunking and IP Contact Center Services: Connecting Cisco Unified Communications Manager 7.1(3) with Cisco Unified Border Element (Enterprise Edition) 1.3

© 2010 Cisco Systems, Inc. All rights reserved.

Important notices, privacy statements, and trademarks of Cisco Systems, Inc. can be found on cisco.com of 49

January 19, 2010

Table of Contents

Introduction..................................................................................................................................................................................................................................................... 3 Verizon IP Trunking Overview....................................................................................................................................................................................................................... 3 Verizon IPCC Overview.................................................................................................................................................................................................................................. 3 Typical Reference Network:............................................................................................................................................................................................................................ 4 System Components........................................................................................................................................................................................................................................ 4

Hardware Components............................................................................................................................................................................................................................... 4 Software Requirements............................................................................................................................................................................................................................... 5

Features............................................................................................................................................................................................................................................................ 5 Features Supported (IP Trunking).............................................................................................................................................................................................................. 5 Features Not Supported (IP Trunking)....................................................................................................................................................................................................... 5 Features Supported (IPCC)........................................................................................................................................................................................................................ 5 Features Not Supported (IPCC).................................................................................................................................................................................................................. 6

Call Flow Overview......................................................................................................................................................................................................................................... 6 Outbound Call Flows.................................................................................................................................................................................................................................. 6

Failover ................................................................................................................................................................................................................................................ 6 Example call flow for Voice Calls (G.729)......................................................................................................................................................................................... 7 Example call flow for FAX Calls (G.711ulaw).................................................................................................................................................................................. 7

Inbound Call Flows.................................................................................................................................................................................................................................... 7 Failover ................................................................................................................................................................................................................................................ 8

Communications Manager Configuration........................................................................................................................................................................................................ 9 Cisco UCM SIP Session Expires Timer..................................................................................................................................................................................................... 9 Early-Media Cut-thru: Enable PRACK on Cisco UCM........................................................................................................................................................................... 10 Route Group Configuration:..................................................................................................................................................................................................................... 17

Route List for FAX: .......................................................................................................................................................................................................................... 20 The previously defined ROUTE GROUP is selected in the FAX Route List (similar to Voice Route List)....................................................................................... 20 Route Pattern for Voice:................................................................................................................................................................................................................... 22 Route Pattern for FAX: .................................................................................................................................................................................................................... 23

EMEA Configuration..................................................................................................................................................................................................................................... 24 EMEA Cisco UCM Configuration........................................................................................................................................................................................................... 24 EMEA CISCO UBE dial-peer Configuration........................................................................................................................................................................................... 28

IPCC Configuration....................................................................................................................................................................................................................................... 30 IPCC CISCO UCM Configuration........................................................................................................................................................................................................... 30 IPCC CISCO UBE dial-peer Configuration............................................................................................................................................................................................. 32

Cisco UBE Example Configuration (North America)................................................................................................................................................................................... 34 Configuration of Cisco Unified Border Element (CISCO UBE) 1.3........................................................................................................................................................ 34

Troubleshooting............................................................................................................................................................................................................................................. 45 References................................................................................................................................................................................................................................................ 46

Acronyms....................................................................................................................................................................................................................................................... 47



IntroductionIntroductionIntroductionIntroduction

This application note describes how to configure a Cisco Unified Communications Manager (Cisco UCM) 7.1 and Cisco Unified Border Element (Cisco UBE)

Enterprise Edition 1.3 for connectivity to Verizon’s IP trunking service. The deployment model covered in this application note utilizes Verizon’s Private IP

(commercial MPLS network) to access Verizon IP Trunking. Supplemental guidelines are also included for using Verizon IP trunking to interface to their IP-based

Contact Center Service or IPCC. Please note that in the context of this document, “IPCC” refers to a cloud-based Contact Center product from Verizon, and should not

be confused with a Cisco product. Additional supplemental guidelines are provided for an EMEA configuration.

Testing was performed in accordance with the test plans for the Verizon IP trunking (United States, Europe, Middle East and Africa), and IP Contact Center services. All

features were verified.

Although this document does not detail the results of the testing performed it provides the essential configurations required for SIP interoperability with Cisco

UCM/Cisco UBE and the Verizon IP Trunking and IPCC services.

Verizon IP TrunkingVerizon IP TrunkingVerizon IP TrunkingVerizon IP Trunking Overview Overview Overview Overview

Verizon IP trunking services simplify management of your network and can help drive operational efficiencies. They do this by consolidating your voice services onto a

SIP-based VoIP network, thereby optimizing your data IP network, and controlling costs associated with maintaining traditional TDM local lines, trunks, and dedicated

PRI circuits. Verizon also offers a native IP Trunking option that provides a SIP trunk directly to your IP PBX, and an IP Integrated Access option that leverages a

gateway device so you can interface with legacy key or PBX systems.

Verizon’s latest Burstable Enterprise Shared Trunking (BEST) feature enhancement allows you to share all your voice trunking resources across your enterprise and lets

you use idle trunk capacity in one location to accommodate a traffic increase in another location. BEST helps control costs, as fewer concurrent calls need to be

purchased at each location and resources can be shared to provide time of day benefits and peak usage management.

Verizon IPCC OverviewVerizon IPCC OverviewVerizon IPCC OverviewVerizon IPCC Overview

Verizon VoIP Inbound is a component of the IP Contact Center (IPCC) portfolio of internetworking services, which tightly couples signaling and functionality from the

Advanced Toll Free and IP networks to deliver the intelligent routing and call treatment required by contact centers. The IPCC services are network-based and include IP

Interactive Voice Response (IVR) in addition to VoIP Inbound.

VoIP Inbound is standards-compliant and provides single-call service that allows PSTN-originated toll free calls to seamlessly terminate and transfer to a SIP or TDM

endpoints, without call re-originations that tie up customer premises equipment (CPE) port capacity. VoIP Inbound includes advanced toll free features -including



automatic ISDN User Part and SIP Error overflow for reliable termination to SIP or TDM devices anywhere; and, when combined with IP IVR, supports customer-

driven pre/post call routing and/or call treatment and queuing for customers using Cisco ICM

Typical Reference Network:Typical Reference Network:Typical Reference Network:Typical Reference Network:

System ComponentsSystem ComponentsSystem ComponentsSystem Components

Hardware Components

• Cisco UBE IOS version 12.4(20)T4. Primary and Secondary Cisco UBE routers are used for high availability.

• Cisco Unified Border Element is an integrated Cisco IOS Software application that runs on various hardware platforms, for more details: http://www.cisco.com/go/cube

• Packet Voice Data Module (PVDM). You will need to install DSP modules on a supported ISR platform if you require MTP, transcoding or conference bridge resources. These DSP resources are co-resident on the CISCO UBE routers in our lab configuration.

• Cisco UCM cluster with (2) Cisco MCS 7800 Series servers (Cisco Unified Communications Manager)

• Cisco Unified IP Phones

• Analog Telephony Adapter for FAX, modem, or analog phones

http://www.cisco.com/go/cube



• Ethernet Switch

• WAN router used to terminate the Verizon MPLS network

Software Requirements

• Cisco Unified Call Manager 7.1.3

• Cisco Unified Border Element Cisco UBE running IOS version 12.4(20)T4

FeaturesFeaturesFeaturesFeatures

Features Supported (IP Trunking)

• Voice calls using G.729 codec

• RFC3261 generic feature Support

• Locating SIP Servers via DNS SRV and DNS A Records

• Early Media Cut-Through

• Calling Party Number Presentation and Privacy (P-Asserted Identity)

• FAX calls using G.711ulaw passthrough

• DTMF as RFC2833 NTE (named telephone events) when a compressed audio codec is used.

Note: RFC2833 is not currently supported when using CTI Route-Points on CISCO UCM 7.1. An MTP resource is required to enable DTMF relay for any calls that utilize a CTI Route-point.

• CISCO UBE performs Delay-Offer-to-Early-Offer interworking of an initial SIP INIVTE from CISCO UCM without SDP

Features Not Supported (IP Trunking)

• T.38 Fax relay is not supported by Verizon IP Trunking Service at this time Note: If you have a Cisco Fax Server or other T.38 Fax device, you will need to ensure that design considerations have been made to support this outside of the Verizon IP Trunking service. (i.e…T1 PRI)

• Mid-call codec negotiation (example: G.729 upspeed to G.711) this capability is not currently supported with CISCO UCM or CISCO UBE.

• Outbound SIP REFER with Replaces. CISCO UCM does not currently support generation of an outbound SIP Refer with replaces messaging.

• CISCO UCM 7.1(3) can only support a single codec between the end device (i.e. IP Phone, ATA) and the SIP Trunk. A workaround for this used during testing was to create multiple Regions and Device pools in order to control the codec selection prior to being presented to the SIP Trunk. The end devices were configured with a specific Device Pool based on the codec used for off-net calls. See configuration section for details



Features Supported (IPCC)

• RFC3261 generic feature Support

• Locating SIP Servers via DNS SRV and DNS A Records

• Early Media Cut-Through

• Calling Party Number Presentation and Privacy (P-Asserted Identity)

• DTMF as RFC2833 NTE (named telephone events) when a compressed audio codec is used

• Cisco UBE performs Delay-Offer-to-Early-Offer interworking of an initial SIP INIVTE from IPCC to Cisco UCM

Features Not Supported (IPCC)

• Outbound SIP REFER with Replaces. Cisco UCM does not currently support generation of an outbound SIP Refer with replaces messaging.

• Due to the codec negotiation issues for certain IPCC call flows, (Enhanced Transfer) it is necessary to configure the DIDs used for incoming IPCC calls for the G.711ulaw codec only. This will allow all calls presented by IPCC to negotiate a single codec (G.711ulaw) and allow proper media flow when using advanced call transfer services.

Call Flow OverviewCall Flow OverviewCall Flow OverviewCall Flow Overview

Outbound Call Flows

The same SIP trunks are utilized between Cisco UCM to Cisco UBE for both Voice and FAX off-net calls. However, the call type (i.e., Voice vs. FAX) must be differentiated to ensure the desired codec is used. This delineation is achieved by performing digit manipulation at the Route List prior to the call being delivered to the Route Group. Each type of device (i.e., IP Phones vs analog devices for FAX) will have separate Route-Patterns that belong to their respective partition. The route patterns will then route the call to the specified Route List. The Route List is used to distinguish a Voice call from a FAX call by manipulating the called party numbers. A voice call is forwarded with a leading 9. FAX calls will strip the leading 9 and prepend the called party number with an 8. After the digit manipulation, the Route List then forwards the call to the Route Group, which routes the call to the SIP trunks. The SIP trunks are the same for ALL calls from Cisco UCM to Cisco UBE (see example call flows below).

Failover

Outbound calls can either be sent to the SIP Trunks in a “Top-Down” or “Round-Robin” method. Regardless of the method used, if when the call gets routed to the Cisco UBE and the Cisco UBE is not able to complete the call , the call is then routed to the next SIP Trunk or Cisco UBE in the Route-group. This provides redundancy for outbound calls by using multiple Cisco UBE devices connecting the Verizon VoIP network.



Example call flow for Voice Calls (G.729)

Example call flow for FAX Calls (G.711ulaw)

Inbound Call Flows

Inbound calls are received from either the IP Trunking or IPCC services. These services provide a variable length user ID for routing the SIP call. The last 4 digits of the User ID are used to route the call within the IP PBX. The IP PBX then connects the call to the corresponding IP Phone or analog device.

Route Pattern 9@ For FAX Calls

Route List

Route Group

CUCM Cluster

CUBE 2

CUBE

CUBE

A VERIZON VoIP

CUBE 1 SIP Trunk

9 is stripped on FAX calls in CUCM and replaced with 8 8 is stripped in CUBE

for FAX calls

Route Pattern 9@ For Voice Calls

Route List

Route Group

CUCM Cluster

CUBE 2

CUBE

CUBE

VERIZON VoIP

CUBE 1 SIP Trunk

No digits stripped on Voice calls in CUCM

9 is stripped in CUBE for Voice calls



Failover

The VoIP Network sends periodic SIP options messages as a keep alive mechanism to determine the state of the Cisco UBE devices.

If the primary Cisco UBE does not respond to these options messages, the calls are then routed to the Secondary Cisco UBE router.

Note: The Cisco UBE will respond to the SIP options pings by default. No additional configuration is necessary.

The VoIP network will also re-route any calls to the secondary CISCO UBE if it receives a temporary call setup failure SIP message from the primary CISCO UBE. (Example: 503 or 404 messages)

To allow failover for inbound calls when the primary CISCO UBE device is unable to contact the CISCO UCM cluster.

In the CISCO UBE:

1. Configure " monitor probe icmp-ping" to any dial-peers connecting to the CISCO UCM cluster.

2. Add " call fallback monitor" to the global configuration

3. Change the PSTN cause code mapping under the SIP-UA configuration " set pstn-cause 1 sip-status 503"

Without this configuration the incoming call setup from the Verizon IP trunking service may time-out and the call would be cancelled before trying the secondary CISCO UBE device.



Communications Manager ConfigurationCommunications Manager ConfigurationCommunications Manager ConfigurationCommunications Manager Configuration

Cisco UCM SIP Session Expires Timer

Cisco UCM will invoke the Session Expires Timer if the SIP Session Timer for all SIP calls. This timer is calculated at seconds/2 and the default value is 1800, with this default timer setting SIP calls may disconnect after 15 Min. As a workaround we set this parameter to the maximum value of 86400 in the CISCO UCM Service Parameters. This allows the SIP call to be active for 12 hours before the CISCO UCM SIP session expires timer engages and disconnects the call.



Cisco UCM System Cluster wide Parameters (SIP)



Early-Media Cut-thru: Enable PRACK on Cisco UCM

Early media refers to media (e.g., audio and video) that is exchanged before the called-party accepts a particular session. Typical examples of early media generated by the called-party are ringing tone and announcements (e.g., queuing status). Early media generated by the caller typically consists of voice commands or dual tone multi-frequency (DTMF) tones to drive interactive voice response (IVR) systems. Enabling PRACK is required in order to allow early media between CISCO UCM and CISCO UBE.

PRACK- Provisional Acknowledgement to a Session not yet established

• Purpose is to acknowledge progress information on a requested process • The INVITE Includes a Require header stipulating the User Agent Client (UAC) wants a reliable provisional response

SIP Rel1XX Enabled: This parameter determines whether all SIP provisional responses (other than 100 Trying messages) get sent reliably to the remote SIP endpoint.

If this parameter is disabled, Cisco CallManager does not acknowledge or confirm 18X messages. Valid values specify True (acknowledge 18X messages with PRACK) or False (do not acknowledge 18X messages with PRACK).

CIisco UCM Administrator>System>Service Parameters

Change the SIP Rel1XX Enabled from the default of False to True



Note: No changes are required on the CISCO UBE. The CISCO UBE supports PRACK and Early Media by default.



Media Resource Group List: List of Media Resource Groups configured for the SIP Trunk MRGL



Media Resource Group: Configured Conference Bridge resource associated with DSP resources configured on CISCO UBE



CODEC Selection using Device Pools and Regions All Voice calls through the SIP trunk should use G.729 and FAX devices should use G.711. Note in the configuration below, there are two regions. Calls between the

“Default” and “Offsite” region will use G.729 and calls between “Default” and “Offsite” use G.711. Applying this configuration to our testbed, the SIP trunk is placed in

a Device Pool with the “Offsite” region, and phone devices should be placed in a Device Pool that with the “Default” region. Devices used for analog FAX should use a

Device Pool with the “Offsite” region. Devices that belong to the same region are configured to use the G.711 codec

Note: With CISCO UCM 7.1 the system defaults for Intra-Region codec preference is to use the highest quality codec. By default this is G722 or G711. The system default for Inter-Region codec preference is G729. The above region configuration is used to ensure that these codecs will be used if the system defaults are changed.



List of Device pools and the associated Regions

List of Phones and ATA Devices: The Device Pools selected determine the codec used for Off-net calls



SIP Trunk Configuration: SIP Trunk Configuration: SIP Trunk Configuration: SIP Trunk Configuration:

The Offsite Device Pool is configured for codec negotiation and the SIP_Trunk_MRGL is selected for Conference Bridge resources.

Note: Note: MTP required Not Selected



Route Group Configuration:

Both SIP Trunks are members of the same Route Group



Route List for Voice: The previously defined ROUTE GROUP is selected in the Voice Route List



Route List Details for Voice: No Digits are discarded for off-net Voice calls. The leading “9” is preserved when the call is forwarded to the CISCO UBE, this allows the CISCO UBE to differentiate the call as voice and use the corresponding G.729 CODEC.



Route List for FAX: The previously defined ROUTE GROUP is selected in the FAX Route List (similar to Voice Route List)



Route List Details for FAX: The “9” is stripped from the called party number and replaced with an “8”.

This dial plan configuration ensures that the user only needs to dial a “9” for Voice and FAX off-net calls.



Route Pattern for Voice: This Route Pattern is in the Voice partition and is serviced by the Voice Route List (No Digits are stripped)



Route Pattern for FAX: This Route Pattern is in the FAX partition and is serviced by the FAX Route List (No Digits are stripped)



EMEA ConfigurationEMEA ConfigurationEMEA ConfigurationEMEA Configuration

EMEA Cisco UCM Configuration

The following steps are required to enable localised Network tones and User Interface:

1. Download necessary localization files from http://www.cisco.com/cisco/web/download/index.html (requires valid CCO account)

2. Install localization software on every Cisco Unified Communications Manager in the cluster.

This does require a restart to enable the localisation file after installation.

http://www.cisco.com/cisco/web/download/index.html



3. Using the CISCO UCM Administration website either change the locale information at the device pool level or at the phone device level.

Example shows change to network locale on the phone configuration page:

Note: User locale changes the user interface only and is controlled independently of the network tones.



4. Verify that all devices (Phones and Gateways) that are in the same Region to allow use of the G.711alaw codec. This is similar to the above configuration for FAX

end-points.

5. Next create a variable-length Route-Pattern with “#” as terminating digit.

Example: 9.011!#

Note: The previously configured Voice Route List is utilized for this route-pattern in order to allow the complete calling number to be sent to CISCO UBE.



EMEA CISCO UBE dial-peer Configuration

The CISCO UBE configuration for EMEA is very similar to the US (Domestic) IP Trunking configuration.

The major difference being that the dial-peers are configured to support G.711alaw.

Note: With EMEA requiring only a single codec, the creation of separate dial-peers for FAX is not required.

dial-peer Voice 100 voip description OUTBOUND G729 Voice SIP calls to VerizonB translation-profile outgoing DIGITSTRIP-9 destination-pattern 9T codec g711alaw session protocol sipv2 session target sip-server dtmf-relay rtp-nte ip qos dscp af32 signaling no vad ! dial-peer Voice 101 voip description INBOUND Voice SIP calls from VerizonB EMEA codec g711alaw session protocol sipv2 session target sip-server incoming called-number [1-5]... dtmf-relay rtp-nte no vad ! dial-peer Voice 102 voip description To/From CUCM subscriber for Voice preference 2 destination-pattern [1-5]... monitor probe icmp-ping codec g711alaw session protocol sipv2 session target ipv4:192.168.0.4 incoming called-number 9T FAX rate disable no vad ! dial-peer Voice 103 voip



description To/From CUCM publisher for Voice preference 5 destination-pattern 1... monitor probe icmp-ping codec g711alaw session protocol sipv2 session target ipv4:192.168.0.6 incoming called-number 9T dtmf-relay rtp-nte no vad



IPCC ConfigurationIPCC ConfigurationIPCC ConfigurationIPCC Configuration

IPCC CISCO UCM Configuration

The CISCO UCM Configuration changes required for IPCC services to work properly are:

1. Verify all IPCC end-points (Phones and Gateways) are in the same Region to allow negotiation of the G.711ulaw codec.

2. Disable diversion-header support on the SIP Trunk device configuration.



3. For out-bound IPCC calls a 9.1800632XXXX Route-pattern must be configured in the Communications Manager.



IPCC CISCO UBE dial-peer Configuration

The CISCO UBE dial-peers must be configured to negotiate only the G.711 codec for all IPCC inbound calls.

This requires specific incoming called numbers for IP Toll-Free calls.

Example: User calls 8005551212 and IPCC routes the call to 1212 with the following dial-peer configured on the CISCO UBE router.

Note: In this example the IPCC network is only sending the last 4 digits of the called number.

dial-peer voice 800 voip description OUTBOUND to VzB IP Toll Free translation-profile outgoing DIGITSTRIP-9 destination-pattern 91800632T codec g711ulaw session protocol sipv2 session target dns:rchtcsd05011.vzbi.com dtmf-relay rtp-nte ip qos dscp af32 signaling no vad ! ! ! dial-peer voice 801 voip description G.711 INBOUND from VzB IP Toll Free codec g711ulaw session protocol sipv2 session target sip-server incoming called-number 1212 dtmf-relay rtp-nte no vad ! ! ! ! dial-peer voice 802 voip description G.711 To/From CUCM subscriber IP Toll Free preference 2 destination-pattern 1212 monitor probe icmp-ping codec g711ulaw session protocol sipv2 session target ipv4:192.168.0.4 dtmf-relay rtp-nte no vad



! ! ! ! dial-peer voice 803 voip description G.711 To/From CUCM publisher IP Toll Free preference 5 destination-pattern 1212 monitor probe icmp-ping voice-class codec 2 voice-class sip early-offer forced session protocol sipv2 session target ipv4:192.168.0.6 dtmf-relay rtp-nte no vad



CCCCisco isco isco isco UBE UBE UBE UBE Example C Example C Example C Example Configurationonfigurationonfigurationonfiguration (North America) (North America) (North America) (North America)

Configuration of Cisco Unified Border Element (CISCO UBE) 1.3

Critical commands are marked in Bold with footnotes at bottom of the page

version 12.4 service timestamps debug datetime msec service timestamps log datetime msec no service password-encryption service sequence-numbers ! hostname cube.1.3825 ! boot-start-marker boot-end-marker ! card type t1 2 1 logging message-counter syslog logging buffered 100000 no logging console enable password cisco ! no aaa new-model no network-clock-participate slot 2 no network-clock-participate wic 0 ! dot11 syslog ip source-route no ip dhcp use vrf connected ip dhcp excluded-address 192.168.0.0 192.168.0.100 ! ! !

ip dhcp pool IPPHONES1

network 192.168.0.0 255.255.255.0 default-router 192.168.0.10 option 150 ip 192.168.0.6 !

1 (Optional ) DHCP Service: automatically assign IP address and TFTP server (option 150) configuration to IP Phones



! ip cef ! !

ip domain name pipiptrunksit2.gsiv.com2

ip name-server 166.38.98.2 ip name-server 10.0.1.4 ! no ipv6 cef multilink bundle-name authenticated ! ! ! Voice-card 0 dspfarm dsp services dspfarm ! Voice-card 2 dspfarm ! ! Voice service voip address-hiding

allow-connections sip to sip3

sip

early-offer forced4

localhost dns:ciscocm7.pipiptrunksit2.gsiv.com5

midcall-signaling passthru6

! ! voice class codec 1 codec preference 1 g729r8

2 DNS Domain name for SIP Realm and name server list for DNS resolution 3 Allow SIP to SIP call processing 4 Use this command to forcefully configure a Cisco Unified Border Element to send a SIP invite with SDP on the Out-Leg (OL), Delayed-Offer to Early-Offer for SIP calls. This is applied to all voip dial-peers. 5 Configures global settings for substituting a DNS localhost name in place of the physical IP address in the From, Call-ID, and Remote-Party-ID headers of outgoing messages. 6 Enables support for SIP Supplementary Services (Only used for SIP-to-SIP Calls)



codec preference 2 g711ulaw codec preference 3 g711alaw ! ! ! ! ! ! ! ! voice translation-rule 8 rule 2 /^8\(.*\)/ /\1/ ! voice translation-rule 9 rule 2 /^9\(.*\)/ /\1/ ! !

voice translation-profile DIGITSTRIP-87

translate called 8 !

voice translation-profile DIGITSTRIP-98

translate called 9 ! ! ! ! ! ! ! ! ! ! username cisco privilege 15 secret cisco archive log config hidekeys ! ! ! !

7 Strip the leading “8” from outgoing called number 8 Strip the leading “9” from outgoing called number



! ! ip ssh version 2 ! ! ! ! interface GigabitEthernet0/0

description CUBE inside interface9

ip address 192.168.0.10 255.255.255.0 ip virtual-reassembly load-interval 30 duplex auto speed auto media-type rj45 ! interface GigabitEthernet0/1 description CUBE outside interface ip address 172.16.2.10 255.255.255.0 ip virtual-reassembly load-interval 600 duplex auto speed auto media-type rj45 ! ip route 0.0.0.0 0.0.0.0 172.16.2.1 no ip http secure-server ! ! ip rtcp report interval 10000 ! ! ! control-plane !

call fallback monitor10

call treatment on call threshold global cpu-avg low 68 high 75 call threshold global total-mem low 75 high 85

9 No SIP bind commands configured, SIP is sourced from both inside and outside interfaces 10 The call fallback monitor command is used as a statistics collector of network conditions based upon probes. This is required to monitor the status of the remote destination of the CISCO UCM dial-peer.



call threshold global total-calls low 9 high 1111

! ! ! !

Voice-port 0/1/012

no non-linear playout-delay maximum 120 playout-delay nominal 15 playout-delay minimum low timeouts interdigit 2 timeouts call-disconnect 3 timing digit 300 caller-id enable ! Voice-port 0/1/1 ! ! sccp local GigabitEthernet0/0 sccp ccm 192.168.0.6 identifier 2 priority 2 version 6.0+ sccp ccm 192.168.0.4 identifier 5 priority 1 version 6.0+ sccp ! sccp ccm group 10 associate ccm 5 priority 1 associate ccm 2 priority 2 associate profile 12 register conf001 associate profile 11 register xcode001 !

dspfarm profile 11 transcode13

codec g711ulaw codec g711alaw codec g729ar8 codec g729abr8 codec g729r8 maximum sessions 10 !

dspfarm profile 12 conference14

11 Global Call Admission Control based on Resource utilization 12 Optional FXS port for FAX devices connected directly to the CISCO UBE 13 DSP Resources for Transcoding registered with CISCO UCM cluster



description conference bridge codec g711ulaw codec g711alaw codec g729ar8 codec g729abr8 codec g729r8 codec g729br8 maximum sessions 10 associate application SCCP ! ! ! dial-peer Voice 10 pots description connection to FXS port for FAX calls preference 1 service session destination-pattern 1017 FAX rate disable port 0/1/0 ! dial-peer Voice 100 voip description OUTBOUND G729 Voice SIP calls to VerizonB

translation-profile outgoing DIGITSTRIP-915

destination-pattern 9T16

Voice-class codec 1 session protocol sipv2 session target sip-server

dtmf-relay rtp-nte17

ip qos dscp af32 signaling no vad ! dial-peer Voice 101 voip description INBOUND G729 Voice SIP calls from VerizonB Voice-class codec 1 session protocol sipv2 session target sip-server

14 DSP resources for Conferencing registered with CISCO UCM cluster 15 Strip the leading “9” from outgoing called number 16 Match on outbound calls from CISCO UCM with leading “9” 17 Forwards DTMF tones by using RTP with the Named Telephone Event (NTE) payload type.



incoming called-number [1-5]...18

dtmf-relay rtp-nte no vad ! dial-peer Voice 102 voip description To/From CUCM subscriber for Voice preference 2 destination-pattern [1-5]... monitor probe icmp-ping Voice-class codec 1 session protocol sipv2 session target ipv4:192.168.0.4

incoming called-number 9T19

FAX rate disable no vad ! dial-peer Voice 103 voip description To/From CUCM publisher for Voice preference 5 destination-pattern 1... monitor probe icmp-ping Voice-class codec 1 session protocol sipv2

session target ipv4:192.168.0.620

incoming called-number 9T dtmf-relay rtp-nte no vad ! dial-peer Voice 200 voip description inbound FAX dial peer from VZ session protocol sipv2 session target sip-server

incoming called-number 101821

18 Enables CISCO UBE to set configuration parameters to incoming calls based on the received called number. This ensures that both legs of the SIP call have matching CODECs 19 Enables CISCO UBE to set configuration parameters to outgoing calls based on the received called number. This ensures that both legs of the SIP call have matching CODECs 20 Multiple SIP trunks configured for redundant connections to the CISCO UCM cluster 21 Match on inbound call-leg for FAX calls. This ensures that both legs of the SIP call have matching CODECs.



codec g711ulaw22

FAX rate disable23

no vad ! dial-peer Voice 201 voip description outbound FAX calls to VZ

translation-profile outgoing DIGITSTRIP-824

destination-pattern 8T25

session protocol sipv2 session target sip-server codec g711ulaw FAX rate disable ip qos dscp af32 signaling no vad ! dial-peer Voice 202 voip description To/From CUCM subscriber for FAX Calls preference 2 destination-pattern 1018

monitor probe icmp-ping26

session protocol sipv2 session target ipv4:192.168.0.4

incoming called-number 8T27

codec g711ulaw FAX rate disable no vad ! ! !

22 CODEC is set on dial-peer to force use of g711ulaw for FAX calls. 23 Disables FAX relay transmission capability. FAX-Passthrough is the supported FAX method. 24 Strip leading “8” from outbound FAX calls before sending to VERIZON 25 Match outbound FAX calls from CISCO UCM cluster with leading “8” 26 Enables monitoring of dial-peer targets using ICMP ping. 27 Enables CISCO UBE to set configuration parameters to outgoing calls based on the received called number. This ensures that both legs of the SIP call have matching CODECs



dial-peer Voice 203 voip description To/From CUCM publisher for FAX Calls preference 5 destination-pattern 1018 monitor probe icmp-ping session protocol sipv2

session target ipv4:192.168.0.628

incoming called-number 8T codec g711ulaw FAX rate disable no vad ! dial-peer voice 800 voip description OUTBOUND to VzB IP Toll Free translation-profile outgoing DIGITSTRIP-9 destination-pattern 91800632T codec g711ulaw session protocol sipv2

session target dns:rchtcsd05011.vzbi.com29

dtmf-relay rtp-nte ip qos dscp af32 signaling no vad ! ! ! dial-peer voice 801 voip description G.711 INBOUND from VzB IP Toll Free codec g711ulaw session protocol sipv2 session target sip-server incoming called-number 1212 dtmf-relay rtp-nte no vad ! ! ! ! dial-peer voice 802 voip description G.711 To/From CUCM subscriber IP Toll Free preference 2

28 Multiple SIP trunks configured for redundant connections to the CISCO UCM cluster 29 A unique SIP server is used to route calls to the IPCC service vs. the IP Trunking service.



destination-pattern 1212 monitor probe icmp-ping codec g711ulaw session protocol sipv2 session target ipv4:192.168.0.4 dtmf-relay rtp-nte no vad ! ! dial-peer voice 803 voip description G.711 To/From CUCM publisher IP Toll Free preference 5 destination-pattern 1212 monitor probe icmp-ping codec g711ulaw voice-class sip early-offer forced session protocol sipv2 session target ipv4:192.168.0.6 dtmf-relay rtp-nte no vad ! ! sip-ua set pstn-cause 1 sip-status 503

set pstn-cause 102 sip-status 50330

retry invite 2 retry bye 2 retry cancel 2

sip-server dns:pcclv1n0005.pipiptrunksit2.gsiv.com31

g729-annexb override ! line con 0 line aux 0 line vty 0 4 ! scheduler allocate 20000 1000 end

30 Sets the value of the SIP status code that is to correspond with the PSTN cause code. 31 SIP Proxy FQDN name for outbound SIP calls to the IP Trunking service



TroubleshootingTroubleshootingTroubleshootingTroubleshooting

Always capture logs by enabling logging buffer: “logging buffered 200000”

Remember to disable the console logging: “no logging console”

Add sequence numbering for debugs: “service sequence-number”

Debug Commands

debug ccsip all

debug voip ccapi inout

debug voip dialpeer inout

Debug transcoding

debug dspfarm all

Show Commands

Show voip rtp connection

Show call active voice brief

Show sip-ua calls



References

Cisco UBE on Cisco.com


CISCO UCM 7x SIP Trunk Documentation:

http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/srnd/7x/trunks.html#wp1044916

Cisco UBE PBX / Service Provider Interoperability

http://www.cisco.com/go/interoperability

Verizon Business IP Trunking Services

http://www.verizonbusiness.com/us/products/voip/trunking/

Early Media and Ringing Tone Generation in the Session Initiation Protocol (SIP)

http://www.ietf.org/rfc/rfc3960.txt


http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/srnd/7x/trunks.html

http://www.cisco.com/g

http://www.verizonbusiness.com/us/products/voip/trunking/

http://www.ietf.org/rfc/rfc3960.txt



AcronymsAcronymsAcronymsAcronyms

Acronym Definitions

SIP Session Initiation Protocol

SCCP Skinny Client Control Protocol

TDM Time Division Multiplexing

CISCO UCM Cisco Unified Communications Manager

CISCO UBE Cisco Unified Border Element

PRACK Provisional Response Acknowledgement

TUI Telephony User Interface



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THE SPECIFICATIONS AND INFORMATION REGARDING THE PRODUCTS IN THIS MANUAL ARE SUBJECT TO CHANGE WITHOUT NOTICE. ALL

STATEMENTS, INFORMATION, AND RECOMMENDATIONS IN THIS MANUAL ARE BELIEVED TO BE ACCURATE BUT ARE PRESENTED WITHOUT

WARRANTY OF ANY KIND, EXPRESS OR IMPLIED. USERS MUST TAKE FULL RESPONSIBILITY FOR THEIR APPLICATION OF ANY PRODUCTS.

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MANUAL, EVEN IF CISCO OR ITS SUPPLIERS HAVE BEEN ADVISED OF THE POSSIBILITY OF SUCH DAMAGE



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